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[Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK


 
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dujinfang at gmail.com
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PostPosted: Mon May 18, 2009 2:34 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.

482 also means loop detected. my client only has one account logged in
only one place, and no proxy, can I take 482 as 200 OK?

Thanks.

from RFC 3261:

"8.2.2.2 Merged Requests

If the request has no tag in the To header field, the UAS core MUST
check the request against ongoing transactions. If the From tag,
Call-ID, and CSeq exactly match those associated with an ongoing
transaction, but the request does not match that transaction (based
on the matching rules in Section 17.2.3), the UAS core SHOULD
generate a 482 (Loop Detected) response and pass it to the server
transaction."


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brian at freeswitch.org
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PostPosted: Mon May 18, 2009 2:39 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

Is this in regards to FreeSWITCH or something else you're writing?

/b

On May 18, 2009, at 2:34 PM, dujinfang wrote:
Quote:
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.

482 also means loop detected. my client only has one account logged in
only one place, and no proxy, can I take 482 as 200 OK?

Thanks.

from RFC 3261:

"8.2.2.2 Merged Requests

If the request has no tag in the To header field, the UAS core MUST
check the request against ongoing transactions. If the From tag,
Call-ID, and CSeq exactly match those associated with an ongoing
transaction, but the request does not match that transaction (based
on the matching rules in Section 17.2.3), the UAS core SHOULD
generate a 482 (Loop Detected) response and pass it to the server
transaction."



Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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dujinfang at gmail.com
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PostPosted: Mon May 18, 2009 6:57 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

Yes, FS(13263) send out 482 request merged to my voip client.

I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below :


recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:32.811280:
REGISTER [url=sip:voip.xxx.com]sip:voip.xxx.com[/url] SIP/2.0
CSeq: 208 REGISTER
Content-Length: 0





recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:38.814237:
REGISTER [url=sip:voip.xxx.com]sip:voip.xxx.com[/url] SIP/2.0
CSeq: 210 REGISTER





recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:48.821027:
REGISTER [url=sip:voip.xxx.com]sip:voip.xxx.com[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKa9b7ba70783b617e9998dc4dd82eb3c5
From: <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>;tag=a9b7ba70783b617e9998dc4dd82eb3c5
To: "cc" <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>
Call-ID: a9b7ba70783b617e9998dc4dd82eb3c5@192.168.1.100 (a9b7ba70783b617e9998dc4dd82eb3c5@192.168.1.100)
CSeq: 214 REGISTER
Contact: <[url=sip:cc@192.168.1.100:3270;rinstance=1242647429]sip:cc@192.168.1.100:3270;rinstance=1242647429[/url]>
max-forwards: 70
expires: 300
Content-Length: 0





recv 629 bytes from udp/[69.131.94.250]:3270 at 12:40:52.841591:
REGISTER [url=sip:voip.xxx.com]sip:voip.xxx.com[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da
From: "cc" <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>;tag=b8c37e33defde51cf91e1e03e51657da
To: "cc" <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>
Call-ID: b8c37e33defde51cf91e1e03e51657da@192.168.1.100 (b8c37e33defde51cf91e1e03e51657da@192.168.1.100)
CSeq: 1 REGISTER
Contact: <[url=sip:cc@192.168.1.100:3270;rinstance=1242650454]sip:cc@192.168.1.100:3270;rinstance=1242650454[/url]>
max-forwards: 70
expires: 300





sent 439 bytes to udp/[69.131.94.250]:3270 at 12:40:52.841767:
SIP/2.0 482 Request merged
Via: SIP/2.0/UDP 192.168.1.100:3270;rport=3270;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da;received=69.131.94.250
From: "cc" <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>;tag=b8c37e33defde51cf91e1e03e51657da
To: "cc" <[url=sip:cc@voip.xxx.com:5090]sip:cc@voip.xxx.com:5090[/url]>;tag=tjgccmtraDHFc
Call-ID: b8c37e33defde51cf91e1e03e51657da@192.168.1.100 (b8c37e33defde51cf91e1e03e51657da@192.168.1.100)
CSeq: 1 REGISTER
Content-Length: 0




And I also noticed the CSeq if not continues, seems it lost some. but why the CSeq so big while the client directly logins to FS without any proxy and I don't think there is a loop?


Anyway, don't know why FS does not respond to REGISTER sometimes. I updated FS to 13374, will see if it happen again.


On May 19, 2009, at 3:37 AM, Brian West wrote:
Quote:
Is this in regards to FreeSWITCH or something else you're writing?

/b

On May 18, 2009, at 2:34 PM, dujinfang wrote:
Quote:
On register, sometimes my voip client got SIP/2.0 482 Request merged
sometimes got 200 ok.

482 also means loop detected. my client only has one account logged in
only one place, and no proxy, can I take 482 as 200 OK?

Thanks.

from RFC 3261:

"8.2.2.2 Merged Requests

If the request has no tag in the To header field, the UAS core MUST
check the request against ongoing transactions. If the From tag,
Call-ID, and CSeq exactly match those associated with an ongoing
transaction, but the request does not match that transaction (based
on the matching rules in Section 17.2.3), the UAS core SHOULD
generate a 482 (Loop Detected) response and pass it to the server
transaction."



Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com








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brian at freeswitch.org
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PostPosted: Mon May 18, 2009 7:06 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

Please show me a pcap file to may email address because I can bet you FS/Sofia is doing it right 99% of the time.

/b

On May 18, 2009, at 6:55 PM, dujinfang wrote:
Quote:
es, FS(13263) send out 482 request merged to my voip client.

I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below :


Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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dujinfang at gmail.com
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PostPosted: Mon May 18, 2009 7:40 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

I believe.

capturing on tshark -i eth1 -w register.pcap udp port 5090


do you have further suggestions on the tshark filter?


Thanks.



On May 19, 2009, at 8:06 AM, Brian West wrote:
Quote:
Please show me a pcap file to may email address because I can bet you FS/Sofia is doing it right 99% of the time.

/b

On May 18, 2009, at 6:55 PM, dujinfang wrote:
Quote:
es, FS(13263) send out 482 request merged to my voip client.

I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below :


Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com








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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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brian at freeswitch.org
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PostPosted: Mon May 18, 2009 7:42 pm    Post subject: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK Reply with quote

I never use tshark to capture... I use tcpdump -s0 -x port 5090 -w file.pcap

/b

On May 18, 2009, at 7:39 PM, dujinfang wrote:
Quote:
I believe.

capturing on tshark -i eth1 -w register.pcap udp port 5090


do you have further suggestions on the tshark filter?


Thanks.


Brian West
brian@freeswitch.org (brian@freeswitch.org)



-- Meet us at ClueCon! http://www.cluecon.com
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